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Mark Spencer ada1e768c4
Fix unexpected behavior in voicemail (bug #1856)
21 years ago
agi Clean agi-sphinx-test (bug #1547) 21 years ago
apps Fix unexpected behavior in voicemail (bug #1856) 21 years ago
astman Enhancements for zaptel+bsd (bug #1781) 21 years ago
cdr x86-64 compile fixes and cleanups 21 years ago
channels When dialing outbound, if we match a peer name, use it (who really cares?) (bug #1850) 21 years ago
codecs x86-64 compile fixes and cleanups 21 years ago
configs Update sample extensions.conf macro-stdexten to include an a extension that sends the user into voicemailmain when * is pressed 21 years ago
contrib Update IAXY firmware to version 16 (provides better timestamps) 21 years ago
db1-ast Add missing queue header file (bug #1626) 21 years ago
doc Merge agent callback login/logoff events and logging (bug #1775) 21 years ago
editline Cleanup leak in editline (bug #1847) 21 years ago
formats Merge rmarchev's vox fixes (bug #1812) 21 years ago
images Version 0.1.12 from FTP 23 years ago
include/asterisk Separate PROCEEDING and PROGRESS 21 years ago
keys Add information for IAX on Free World Dialup 21 years ago
pbx Cleanup formatting 21 years ago
redhat Add the SuSE AMD64 support and fixes from Bug #706 22 years ago
res x86-64 compile fixes and cleanups 21 years ago
sounds Merge fabeman's extensive meetme improvements (bug #1624) 21 years ago
stdtime Merge FreeBSD locking fixes (bug #1411) 21 years ago
utils Update default astman types for newer newt (bug #1578) 21 years ago
.cvsignore Add support for E1 E&M 21 years ago
BUGS Fix BUGS document to report bug tracker 22 years ago
CHANGES Update changelog 21 years ago
CREDITS add ww 22 years ago
HARDWARE Update HARDWARE 22 years ago
LICENSE Version 0.1.1 from FTP 26 years ago
Makefile x86-64 compile fixes and cleanups 21 years ago
README Fix 2 typos in README 22 years ago
SECURITY Update security document, work on threading with pbx.c and small SIP fixes 21 years ago
acl.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
aescrypt.c Add AES support 22 years ago
aeskey.c Add AES support 22 years ago
aesopt.h Fix AES for MacOS build 21 years ago
aestab.c Add AES support 22 years ago
alaw.c Version 0.1.10 from FTP 24 years ago
app.c Fix unexpected behavior in voicemail (bug #1856) 21 years ago
ast_expr.y More expression fixes (bug #1548 again) 21 years ago
astconf.h Version 0.3.0 from FTP 23 years ago
asterisk.c Formatting fix. Tabs not spaces :) 21 years ago
asterisk.h Have a contact line in responses, merge logging patches 22 years ago
astmm.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
autoservice.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
callerid.c Change strlen calls to ast_strlen_zero in callerid.c 21 years ago
cdr.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
channel.c Separate PROCEEDING and PROGRESS 21 years ago
chanvars.c Include fixes for portability 22 years ago
cli.c Add "show channels concise" for machine readable format 21 years ago
coef_in.h Version 0.1.7 from FTP 25 years ago
coef_out.h Version 0.1.7 from FTP 25 years ago
config.c Add "LOW_MEMORY" option and increase some buffer sizes (bug #1827) 21 years ago
db.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
dlfcn.c Make it build and run on MacOS X 22 years ago
dns.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
dsp.c Make faxdetect configurable and turn OFF by default 21 years ago
ecdisa.h Version 0.1.10 from FTP 24 years ago
enum.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
file.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
frame.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
fskmodem.c Version 0.1.10 from FTP 24 years ago
image.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
indications.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
io.c Make it build and run on MacOS X 22 years ago
loader.c Merge res_odbc and res_config 21 years ago
logger.c Eliminate duplicate init_logger_chain call (bug #1853) 21 years ago
make_build_h Version 0.1.8 from FTP 24 years ago
manager.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
md5.c Fix AES for MacOS build 21 years ago
mkdep Make mkdep throw away stderr since people think the error messages printed are serious when they are not 21 years ago
muted.c Make muted handle transfers 21 years ago
muted.conf.sample clean up config file sample 21 years ago
pbx.c x86-64 compile fixes and cleanups 21 years ago
poll.c Make it build and run on MacOS X 22 years ago
privacy.c Use ast_strlen_zero in privacy.c 21 years ago
rtp.c Handle binding more nicely 21 years ago
sample.call Add example of using Account in sample.call file 21 years ago
say.c Update coding guidelines, fix "say.c" compile on older compilers, update coding guidelines (includes bug #1631) 21 years ago
sched.c Unlock while processing schedule queue 22 years ago
sounds.txt Merge fabeman's extensive meetme improvements (bug #1624) 21 years ago
srv.c More strlen_zero checks (bug #1549) 21 years ago
tdd.c Version 0.1.10 from FTP 24 years ago
term.c Merge Tilghman's color patches for the asterisk prompt (bug #1535) 21 years ago
translate.c Merge FreeBSD locking fixes (bug #1411) 21 years ago
ulaw.c Version 0.1.10 from FTP 24 years ago
utils.c Fix OpenBSD compile (bug #1830) 21 years ago

README

The Asterisk Open Source PBX
by Mark Spencer <markster@digium.com>
Copyright (C) 2001-2004 Digium
================================================================
* SECURITY
  It is imperative that you read and fully understand the contents of
  the SECURITY file before you attempt to configure an Asterisk server.

* WHAT IS ASTERISK
  Asterisk is an Open Source PBX and telephony toolkit.  It is, in a
sense, middleware between Internet and telephony channels on the bottom,
and Internet and telephony applications at the top.  For more information
on the project itself, please visit the Asterisk home page at:

           http://www.asterisk.org

In addition you'll find lot's of information compiled by the Asterisk
community on this Wiki:

           http://www.voip-info.org/wiki-Asterisk

* LICENSING
  Asterisk is distributed under GNU General Public License.  The GPL also
must apply to all loadable modules as well, except as defined below.

  Digium, Inc. (formerly Linux Support Services) retains copyright to all 
of the core Asterisk system, and therefore can grant, at its sole discretion, 
the ability for companies, individuals, or organizations to create proprietary
or Open Source (but non-GPL'd) modules which may be dynamically linked at
runtime with the portions of Asterisk which fall under our copyright
umbrella, or are distributed under more flexible licenses than GPL.  


  If you wish to use our code in other GPL programs, don't worry -- there
is no requirement that you provide the same exemption in your GPL'd
products (although if you've written a module for Asterisk we would
strongly encourage you to make the same exemption that we do).

  Specific permission is also granted to OpenSSL and OpenH323 to link to
Asterisk.

  If you have any questions, whatsoever, regarding our licensing policy,
please contact us.

  Modules that are GPL-licensed and not available under Digium's 
licensing scheme are added to the Asterisk-addons CVS module.
  
* REQUIRED COMPONENTS

== Linux ==
  Currently, the Asterisk Open Source PBX is only known to run on the
Linux OS, although it may be portable to other UNIX-like operating systems
(like FreeBSD) as well. 


* GETTING STARTED

First, be sure you've got supported hardware (but note that you don't need ANY hardware, not even a soundcard) to install and run Asterisk. Supported are:

	* All Wildcard (tm) products from Digium (www.digium.com)
	* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
	* Full Duplex Sound Card supported by Linux
	* Adtran Atlas 800 Plus
	* ISDN4Linux compatible ISDN card
	* Tormenta Dual T1 card (www.bsdtelephony.com.mx)

Hint: CAPI compatible ISDN cards can be run using the add-on channel chan_capi.

So let's proceed:

1) Run "make"
2) Run "make install"

If this is your first time working with Asterisk, you may wish to install
the sample PBX, with demonstration extensions, etc.  If so, run:

	"make samples"

Doing so will overwrite any existing config files you have. If you are lacking a soundcard you won't be able to use the DIAL command on the console, though.

Finally, you can launch Asterisk with:

	./asterisk -vvvc

You'll see a bunch of verbose messages fly by your screen as Asterisk
initializes (that's the "very very verbose" mode).  When it's ready, if
you specified the "c" then you'll get a command line console, that looks
like this:

*CLI>

You can type "help" at any time to get help with the system.  For help
with a specific command, type "help <command>".  To start the PBX using
your sound card, you can type "dial" to dial the PBX.  Then you can use
"answer", "hangup", and "dial" to simulate the actions of a telephone.
Remember that if you don't have a full duplex sound card (And asterisk
will tell you somewhere in its verbose messages if you do/don't) than it
won't work right (not yet).

Feel free to look over the configuration files in /etc/asterisk, where
you'll find a lot of information about what you can do with Asterisk.

* ABOUT CONFIGURATION FILES

All Asterisk configuration files share a common format.  Comments are
delimited by ';' (since '#' of course, being a DTMF digit, may occur in
many places).  A configuration file is divided into sections whose names
appear in []'s.  Each section typically contains two types of statements,
those of the form 'variable = value', and those of the form 'object =>
parameters'.  Internally the use of '=' and '=>' is exactly the same, so 
they're used only to help make the configuration file easier to
understand, and do not affect how it is actually parsed.

Entries of the form 'variable=value' set the value of some parameter in
asterisk.  For example, in tormenta.conf, one might specify:

	switchtype=national

In order to indicate to Asterisk that the switch they are connecting to is
of the type "national".  In general, the parameter will apply to
instantiations which occur below its specification.  For example, if the
configuration file read:

	switchtype = national
	channel => 1-4
	channel => 10-12
	switchtype = dms100
	channel => 25-47

Then, the "national" switchtype would be applied to channels one through
four and channels 10 through 12, whereas the "dms100" switchtype would
apply to channels 25 through 47.
  
The "object => parameters" instantiates an object with the given
parameters.  For example, the line "channel => 25-47" creates objects for
the channels 25 through 47 of the tormenta card, obtaining the settings
from the variables specified above.

* MORE INFORMATION

See the doc directory for more documentation.

Finally, you may wish to visit the web site and join the mailing list if
you're interested in getting more information.

   http://www.asterisk.org/index.php?menu=support

Welcome to the growing worldwide community of Asterisk users!

Mark Spencer